SIP (Session Initiation Protocol) is a signaling protocol used for initiating, managing, and terminating real-time communication sessions over the Internet.
A SIP client is a software application or hardware device that initiates SIP requests to establish media sessions with other SIP clients.
SIP clients allow users to make voice and video calls, send instant messages, and share files and documents over IP networks.
Some common examples of SIP clients include softphones, IP phones, SIP-enabled mobile apps, and SIP-based video conferencing systems. SIP clients must be registered with a SIP server or IP PBX to connect calls and enable communication.
This article provides an overview of SIP clients, their features, working, registration process, and commonly asked questions.
Some key characteristics of a SIP client:
- It is a user agent that contains both a user agent client (UAC) and a user agent server (UAS) component for SIP signaling.
- It can encode and decode audio/video using codecs like G.711, H.264, etc.
- It can manage SIP sessions by generating and parsing SIP requests and responses.
- It registers with a SIP registrar server using SIP REGISTER messages to authenticate the user.
- It supports SIP URI addressing to identify users to call like sip:email@example.com
- It can manage presence subscriptions and notifications using SIP SUBSCRIBE and NOTIFY.
- It can establish media sessions directly with other clients or through SIP proxies.
- It provides a user interface for making and managing calls and SIP account configuration.
- Examples include software phones, VoIP apps, IP phones, SIP ATAs, softswitches, and more.
In summary, a SIP client is the end-user application that enables real-time communications over IP networks using the SIP protocol for signaling and session management.
It combines both signaling and media capabilities.
Types of SIP Clients
There are different types of SIP clients:
1. Software Clients
- Softphones: These are software applications installed on desktops or laptops that enable VoIP calling capabilities. Popular softphones include ZoiPer, Bria, Jitsi, X-Lite, etc.
- Mobile VoIP Apps: Apps like WhatsApp, FaceTime, Google Voice, etc use SIP and VoIP technology for calling features. They are installed on smartphones.
- SIP SDKs: Developer SDKs like PJSIP allow building customized SIP clients integrated into other apps.
2. Hardware Clients
- SIP Phones: Dedicated VoIP desk phones that connect to the internet to make SIP calls. Polycom, Yealink, Cisco, etc are leading vendors.
- SIP ATAs: Analog telephone adapters that connect analog phones to VoIP networks. Popular brands are Obihai, Cisco, and Grandstream.
- IP PBX Systems: On-premise PBX systems like Asterisk, and 3CX with inbuilt SIP client capabilities.
3. Web Clients
- WebRTC Clients: Browser-based SIP clients that work within web pages using WebRTC technology. No downloads are needed.
- Web Phone: VoIP calling interfaces accessible directly through a web browser. Require browser plugins.
Overall, the diversity of SIP clients allows VoIP calling to be integrated into desktop, mobile, web applications, and custom hardware devices.
Developers have ample scope to build tailored SIP solutions.
Features of SIP Clients
1. Registration and Authentication
- SIP clients allow users to register their SIP credentials like username, password, and SIP domains with an SIP registrar server. This enables the server to authenticate the user and maintain their presence information.
- Users can register with multiple SIP domains and maintain multiple SIP identities.
- SIP uses digest authentication to validate the credentials during registration.
2. Presence Management
- SIP clients can subscribe to the presence status of other users on the network. This allows them to see if the user is online, offline, busy, etc.
- The presence status is notified to watchers through SIP NOTIFY messages. This enables real-time presence awareness.
3. Instant Messaging and File Transfer
- SIP clients support instant messaging between users through the SIP MESSAGE method. Users can exchange text messages in real time.
- File transfer is also possible between SIP clients through the REFER method by transferring file metadata.
4. Voice and Video Calling
- The core function of SIP clients is to make VoIP calls by managing SIP sessions. Audio codecs like G.711, G.729, etc enable high voice quality.
- Video calling is enabled by adding video codecs like H.264, and VP8 and using SIP to manage the video session along with audio.
5. Advanced Features
- Additional features offered include call hold, call transfer, call forwarding, call waiting, voicemail, conferencing, etc.
- SIMPLE protocol is used for instant messaging and presence capabilities. WebRTC support is also increasingly common.
- SIP supports TLS and SRTP protocols to enable secure calls and prevent eavesdropping.
- Other security measures include authentication, encryption, and integrity checks on signaling and media.
- SIP allows clients to interact with other SIP-based clients and devices. This enables unified communications.
- Adhering to SIP standards allows for multi-vendor interoperability between clients and servers.
SIP clients provide a robust set of features like call management, instant messaging, presence, security, and interoperability.
These make interactive communications possible directly from end-user applications. The flexibility and ubiquity of SIP fuel the capabilities of modern VoIP clients.
Working with a SIP Client
When a SIP client initiates a call, it goes through these steps:
- Registration – The SIP client registers its SIP URI (e.g. firstname.lastname@example.org) and location with the SIP registrar server. This allows the network to locate the user.
- Call initiation – When the user makes a call, the SIP client sends an INVITE message to the SIP server containing details like the recipient’s address, media capabilities, etc.
- Call management – The SIP messages are exchanged between the caller, callee, and servers to negotiate media capabilities, establish the session, and manage the call flow.
- Media transfer – Once the call is established, RTP (Real Time Protocol) is used to transfer the media (voice/video) between the SIP clients.
- Call termination – When a user ends the call, a SIP BYE message is sent by the client to terminate the session.
For a SIP client to make and receive calls, it must register with a SIP registrar server. The registration process involves:
- The client sends a REGISTER request to the registrar containing its SIP URI and contact address.
- The registrar authenticates the request and stores the user’s SIP URI and IP address in its location service.
- It finds the callee’s contact address when an INVITE request is received.
- The registrar sends a 200 OK response to the client confirming successful registration.
- Registration is periodically refreshed using the Expires header field until the client unregisters.
This allows the SIP network to know the client’s location so it can deliver incoming calls.
SIP clients are allowed to connect to IP networks for affordable voice and video communication, replacing traditional PSTN networks.
They come in various forms like softphones, VoIP desk phones, mobile apps, ATAs, etc., and support standard calling features over an IP network.
Understanding their working and registration process enables configuring and troubleshooting issues in SIP implementations.
Frequently Asked Questions (FAQ)
Ques 1. What is the difference between an SIP account and an SIP client?
Ans. A SIP account refers to the credentials (username, password, SIP URI) used for authentication and identification on a SIP network.
A SIP client is the software or hardware that uses these credentials to register and make VoIP calls over SIP. You can configure multiple SIP accounts on a single SIP client.
Ques 2. What protocols do SIP clients use?
Ans. SIP clients use protocols like SIP, SDP, RTP/RTCP, and TLS/SRTP for signaling, media transfer, and security.
SIP is used for call setup and teardown, SDP for codec and media negotiation, RTP/RTCP for audio/video transfer, TLS provides encryption, and SRTP secure media transfer.
Ques 3. How do SIP clients handle NAT traversal?
Ans. SIP clients behind NAT routers use techniques like STUN, TURN, and ICE to enable communication.
STUN determines the public IP and ports. TURN relays media when direct connectivity fails. ICE finds the optimal path for media traversal across NAT.
Ques 4. What is a softphone?
Ans. A softphone is a software application emulating a hardware phone that enables making VoIP calls directly from a computer.
It requires a headset, microphone, and speakers to work. Popular softphones include Zoiper, Bria, and Linphone.
Ques 5. What are some popular open-source SIP clients?
Ans. Some popular open-source SIP clients are Linphone, Jitsi, Ekiga, and Twinkle. They are free to use and have an open-source codebase that is publicly accessible for customization.
They support standard SIP features for calling, messaging, and conferencing.